freebsd-nq/sys/dev/sound/macio/davbus.c

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/*-
* Copyright 2008 by Marco Trillo. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* $FreeBSD$
*/
/*
* Apple DAVbus audio controller.
*/
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/bus.h>
#include <sys/kernel.h>
#include <sys/lock.h>
#include <sys/malloc.h>
#include <sys/module.h>
#include <sys/mutex.h>
#include <sys/rman.h>
#include <dev/ofw/ofw_bus.h>
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
#include <dev/sound/macio/aoa.h>
#include <dev/sound/macio/davbusreg.h>
#include <machine/intr_machdep.h>
#include <machine/resource.h>
#include <machine/bus.h>
#include "mixer_if.h"
struct davbus_softc {
struct aoa_softc aoa;
phandle_t node;
phandle_t soundnode;
struct resource *reg;
struct mtx mutex;
int device_id;
u_int output_mask;
u_int (*read_status)(struct davbus_softc *, u_int);
void (*set_outputs)(struct davbus_softc *, u_int);
};
static int davbus_probe(device_t);
static int davbus_attach(device_t);
static void davbus_cint(void *);
static device_method_t pcm_davbus_methods[] = {
/* Device interface. */
DEVMETHOD(device_probe, davbus_probe),
DEVMETHOD(device_attach, davbus_attach),
{ 0, 0 }
};
static driver_t pcm_davbus_driver = {
"pcm",
pcm_davbus_methods,
PCM_SOFTC_SIZE
};
DRIVER_MODULE(pcm_davbus, macio, pcm_davbus_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(pcm_davbus, sound, SOUND_MINVER, SOUND_PREFVER, SOUND_MAXVER);
/*****************************************************************************
Probe and attachment routines.
*****************************************************************************/
static int
davbus_probe(device_t self)
{
const char *name;
name = ofw_bus_get_name(self);
if (!name)
return (ENXIO);
if (strcmp(name, "davbus") != 0)
return (ENXIO);
device_set_desc(self, "Apple DAVBus Audio Controller");
return (0);
}
/*
* Burgundy codec control
*/
static int burgundy_init(struct snd_mixer *m);
static int burgundy_uninit(struct snd_mixer *m);
static int burgundy_reinit(struct snd_mixer *m);
static void burgundy_write_locked(struct davbus_softc *, u_int, u_int);
static void burgundy_set_outputs(struct davbus_softc *d, u_int mask);
static u_int burgundy_read_status(struct davbus_softc *d, u_int status);
static int burgundy_set(struct snd_mixer *m, unsigned dev, unsigned left,
unsigned right);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t burgundy_setrecsrc(struct snd_mixer *m, u_int32_t src);
static kobj_method_t burgundy_mixer_methods[] = {
KOBJMETHOD(mixer_init, burgundy_init),
KOBJMETHOD(mixer_uninit, burgundy_uninit),
KOBJMETHOD(mixer_reinit, burgundy_reinit),
KOBJMETHOD(mixer_set, burgundy_set),
KOBJMETHOD(mixer_setrecsrc, burgundy_setrecsrc),
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
KOBJMETHOD_END
};
MIXER_DECLARE(burgundy_mixer);
static int
burgundy_init(struct snd_mixer *m)
{
struct davbus_softc *d;
d = mix_getdevinfo(m);
d->read_status = burgundy_read_status;
d->set_outputs = burgundy_set_outputs;
/*
* We configure the Burgundy codec as follows:
*
* o Input subframe 0 is connected to input digital
* stream A (ISA).
* o Stream A (ISA) is mixed in mixer 2 (MIX2).
* o Output of mixer 2 (MIX2) is routed to output sources
* OS0 and OS1 which can be converted to analog.
*
*/
mtx_lock(&d->mutex);
burgundy_write_locked(d, 0x16700, 0x40);
burgundy_write_locked(d, BURGUNDY_MIX0_REG, 0);
burgundy_write_locked(d, BURGUNDY_MIX1_REG, 0);
burgundy_write_locked(d, BURGUNDY_MIX2_REG, BURGUNDY_MIX_ISA);
burgundy_write_locked(d, BURGUNDY_MIX3_REG, 0);
burgundy_write_locked(d, BURGUNDY_OS_REG, BURGUNDY_OS0_MIX2 |
BURGUNDY_OS1_MIX2);
burgundy_write_locked(d, BURGUNDY_SDIN_REG, BURGUNDY_ISA_SF0);
/* Set several digital scalers to unity gain. */
burgundy_write_locked(d, BURGUNDY_MXS2L_REG, BURGUNDY_MXS_UNITY);
burgundy_write_locked(d, BURGUNDY_MXS2R_REG, BURGUNDY_MXS_UNITY);
burgundy_write_locked(d, BURGUNDY_OSS0L_REG, BURGUNDY_OSS_UNITY);
burgundy_write_locked(d, BURGUNDY_OSS0R_REG, BURGUNDY_OSS_UNITY);
burgundy_write_locked(d, BURGUNDY_OSS1L_REG, BURGUNDY_OSS_UNITY);
burgundy_write_locked(d, BURGUNDY_OSS1R_REG, BURGUNDY_OSS_UNITY);
burgundy_write_locked(d, BURGUNDY_ISSAL_REG, BURGUNDY_ISS_UNITY);
burgundy_write_locked(d, BURGUNDY_ISSAR_REG, BURGUNDY_ISS_UNITY);
burgundy_set_outputs(d, burgundy_read_status(d,
bus_read_4(d->reg, DAVBUS_CODEC_STATUS)));
mtx_unlock(&d->mutex);
mix_setdevs(m, SOUND_MASK_VOLUME);
return (0);
}
static int
burgundy_uninit(struct snd_mixer *m)
{
return (0);
}
static int
burgundy_reinit(struct snd_mixer *m)
{
return (0);
}
static void
burgundy_write_locked(struct davbus_softc *d, u_int reg, u_int val)
{
u_int size, addr, offset, data, i;
size = (reg & 0x00FF0000) >> 16;
addr = (reg & 0x0000FF00) >> 8;
offset = reg & 0xFF;
for (i = offset; i < offset + size; ++i) {
data = BURGUNDY_CTRL_WRITE | (addr << 12) |
((size + offset - 1) << 10) | (i << 8) | (val & 0xFF);
if (i == offset)
data |= BURGUNDY_CTRL_RESET;
bus_write_4(d->reg, DAVBUS_CODEC_CTRL, data);
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) &
DAVBUS_CODEC_BUSY)
DELAY(1);
val >>= 8; /* next byte. */
}
}
/* Must be called with d->mutex held. */
static void
burgundy_set_outputs(struct davbus_softc *d, u_int mask)
{
u_int x = 0;
if (mask == d->output_mask)
return;
/*
* Bordeaux card wirings:
* Port 15: RCA out
* Port 16: Minijack out
* Port 17: Internal speaker
*
* B&W G3 wirings:
* Port 14: Minijack out
* Port 17: Internal speaker
*/
DPRINTF(("Enabled outputs:"));
if (mask & (1 << 0)) {
DPRINTF((" SPEAKER"));
x |= BURGUNDY_P17M_EN;
}
if (mask & (1 << 1)) {
DPRINTF((" HEADPHONES"));
x |= BURGUNDY_P14L_EN | BURGUNDY_P14R_EN;
}
DPRINTF(("\n"));
burgundy_write_locked(d, BURGUNDY_MUTE_REG, x);
d->output_mask = mask;
}
static u_int
burgundy_read_status(struct davbus_softc *d, u_int status)
{
if (status & 0x4)
return (1 << 1);
else
return (1 << 0);
}
static int
burgundy_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
struct davbus_softc *d;
int lval, rval;
lval = ((100 - left) * 15 / 100) & 0xf;
rval = ((100 - right) * 15 / 100) & 0xf;
DPRINTF(("volume %d %d\n", lval, rval));
d = mix_getdevinfo(m);
switch (dev) {
case SOUND_MIXER_VOLUME:
mtx_lock(&d->mutex);
burgundy_write_locked(d, BURGUNDY_OL13_REG, lval);
burgundy_write_locked(d, BURGUNDY_OL14_REG, (rval << 4) | lval);
burgundy_write_locked(d, BURGUNDY_OL15_REG, (rval << 4) | lval);
burgundy_write_locked(d, BURGUNDY_OL16_REG, (rval << 4) | lval);
burgundy_write_locked(d, BURGUNDY_OL17_REG, lval);
mtx_unlock(&d->mutex);
return (left | (right << 8));
}
return (0);
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
burgundy_setrecsrc(struct snd_mixer *m, u_int32_t src)
{
return (0);
}
/*
* Screamer Codec Control
*/
static int screamer_init(struct snd_mixer *m);
static int screamer_uninit(struct snd_mixer *m);
static int screamer_reinit(struct snd_mixer *m);
static void screamer_write_locked(struct davbus_softc *, u_int, u_int);
static void screamer_set_outputs(struct davbus_softc *d, u_int mask);
static u_int screamer_read_status(struct davbus_softc *d, u_int status);
static int screamer_set(struct snd_mixer *m, unsigned dev, unsigned left,
unsigned right);
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t screamer_setrecsrc(struct snd_mixer *m, u_int32_t src);
static kobj_method_t screamer_mixer_methods[] = {
KOBJMETHOD(mixer_init, screamer_init),
KOBJMETHOD(mixer_uninit, screamer_uninit),
KOBJMETHOD(mixer_reinit, screamer_reinit),
KOBJMETHOD(mixer_set, screamer_set),
KOBJMETHOD(mixer_setrecsrc, screamer_setrecsrc),
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
KOBJMETHOD_END
};
MIXER_DECLARE(screamer_mixer);
static int
screamer_init(struct snd_mixer *m)
{
struct davbus_softc *d;
d = mix_getdevinfo(m);
d->read_status = screamer_read_status;
d->set_outputs = screamer_set_outputs;
mtx_lock(&d->mutex);
screamer_write_locked(d, SCREAMER_CODEC_ADDR0, SCREAMER_INPUT_CD |
SCREAMER_DEFAULT_CD_GAIN);
screamer_set_outputs(d, screamer_read_status(d,
bus_read_4(d->reg, DAVBUS_CODEC_STATUS)));
screamer_write_locked(d, SCREAMER_CODEC_ADDR2, 0);
screamer_write_locked(d, SCREAMER_CODEC_ADDR4, 0);
screamer_write_locked(d, SCREAMER_CODEC_ADDR5, 0);
screamer_write_locked(d, SCREAMER_CODEC_ADDR6, 0);
mtx_unlock(&d->mutex);
mix_setdevs(m, SOUND_MASK_VOLUME);
return (0);
}
static int
screamer_uninit(struct snd_mixer *m)
{
return (0);
}
static int
screamer_reinit(struct snd_mixer *m)
{
return (0);
}
static void
screamer_write_locked(struct davbus_softc *d, u_int reg, u_int val)
{
u_int x;
KASSERT(val == (val & 0xfff), ("bad val"));
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) & DAVBUS_CODEC_BUSY)
DELAY(100);
x = reg;
x |= SCREAMER_CODEC_EMSEL0;
x |= val;
bus_write_4(d->reg, DAVBUS_CODEC_CTRL, x);
while (bus_read_4(d->reg, DAVBUS_CODEC_CTRL) & DAVBUS_CODEC_BUSY)
DELAY(100);
}
/* Must be called with d->mutex held. */
static void
screamer_set_outputs(struct davbus_softc *d, u_int mask)
{
u_int x;
if (mask == d->output_mask) {
return;
}
x = SCREAMER_MUTE_SPEAKER | SCREAMER_MUTE_HEADPHONES;
DPRINTF(("Enabled outputs: "));
if (mask & (1 << 0)) {
DPRINTF(("SPEAKER "));
x &= ~SCREAMER_MUTE_SPEAKER;
}
if (mask & (1 << 1)) {
DPRINTF(("HEADPHONES "));
x &= ~SCREAMER_MUTE_HEADPHONES;
}
DPRINTF(("\n"));
if (d->device_id == 5 || d->device_id == 11) {
DPRINTF(("Enabling programmable output.\n"));
x |= SCREAMER_PROG_OUTPUT0;
}
if (d->device_id == 8 || d->device_id == 11) {
x &= ~SCREAMER_MUTE_SPEAKER;
if (mask & (1 << 0))
x |= SCREAMER_PROG_OUTPUT1; /* enable speaker. */
}
screamer_write_locked(d, SCREAMER_CODEC_ADDR1, x);
d->output_mask = mask;
}
static u_int
screamer_read_status(struct davbus_softc *d, u_int status)
{
int headphones;
switch (d->device_id) {
case 5: /* Sawtooth */
headphones = (status & 0x4);
break;
case 8:
case 11: /* iMac DV */
/* The iMac DV has 2 headphone outputs. */
headphones = (status & 0x7);
break;
default:
headphones = (status & 0x8);
}
if (headphones)
return (1 << 1);
else
return (1 << 0);
}
static int
screamer_set(struct snd_mixer *m, unsigned dev, unsigned left, unsigned right)
{
struct davbus_softc *d;
int lval, rval;
lval = ((100 - left) * 15 / 100) & 0xf;
rval = ((100 - right) * 15 / 100) & 0xf;
DPRINTF(("volume %d %d\n", lval, rval));
d = mix_getdevinfo(m);
switch (dev) {
case SOUND_MIXER_VOLUME:
mtx_lock(&d->mutex);
screamer_write_locked(d, SCREAMER_CODEC_ADDR2, (lval << 6) |
rval);
screamer_write_locked(d, SCREAMER_CODEC_ADDR4, (lval << 6) |
rval);
mtx_unlock(&d->mutex);
return (left | (right << 8));
}
return (0);
}
Sound Mega-commit. Expect further cleanup until code freeze. For a slightly thorough explaination, please refer to [1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html . Summary of changes includes: 1 Volume Per-Channel (vpc). Provides private / standalone volume control unique per-stream pcm channel without touching master volume / pcm. Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for backwards compatibility, SOUND_MIXER_PCM through the opened dsp device instead of /dev/mixer. Special "bypass" mode is enabled through /dev/mixer which will automatically detect if the adjustment is made through /dev/mixer and forward its request to this private volume controller. Changes to this volume object will not interfere with other channels. Requirements: - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which require specific application modifications (preferred). - No modifications required for using bypass mode, so applications like mplayer or xmms should work out of the box. Kernel hints: - hint.pcm.%d.vpc (0 = disable vpc). Kernel sysctls: - hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer bypass mode. - hw.snd.vpc_autoreset (default: 1). By default, closing/opening /dev/dsp will reset the volume back to 0 db gain/attenuation. Setting this to 0 will preserve its settings across device closing/opening. - hw.snd.vpc_reset (default: 0). Panic/reset button to reset all volume settings back to 0 db. - hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value. 2 High quality fixed-point Bandlimited SINC sampling rate converter, based on Julius O'Smith's Digital Audio Resampling - http://ccrma.stanford.edu/~jos/resample/. It includes a filter design script written in awk (the clumsiest joke I've ever written) - 100% 32bit fixed-point, 64bit accumulator. - Possibly among the fastest (if not fastest) of its kind. - Resampling quality is tunable, either runtime or during kernel compilation (FEEDER_RATE_PRESETS). - Quality can be further customized during kernel compilation by defining FEEDER_RATE_PRESETS in /etc/make.conf. Kernel sysctls: - hw.snd.feeder_rate_quality. 0 - Zero-order Hold (ZOH). Fastest, bad quality. 1 - Linear Interpolation (LINEAR). Slightly slower than ZOH, better quality but still does not eliminate aliasing. 2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC quality always start from 2 and above. Rough quality comparisons: - http://people.freebsd.org/~ariff/z_comparison/ 3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be directly fed into the hardware. 4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf. 5 Transparent/Adaptive Virtual Channel. Now you don't have to disable vchans in order to make digital format pass through. It also makes vchans more dynamic by choosing a better format/rate among all the concurrent streams, which means that dev.pcm.X.play.vchanformat/rate becomes sort of optional. 6 Exclusive Stream, with special open() mode O_EXCL. This will "mute" other concurrent vchan streams and only allow a single channel with O_EXCL set to keep producing sound. Other Changes: * most feeder_* stuffs are compilable in userland. Let's not speculate whether we should go all out for it (save that for FreeBSD 16.0-RELEASE). * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org> * pull out channel mixing logic out of vchan.c and create its own feeder_mixer for world justice. * various refactoring here and there, for good or bad. * activation of few more OSSv4 ioctls() (see [1] above). * opt_snd.h for possible compile time configuration: (mostly for debugging purposes, don't try these at home) SND_DEBUG SND_DIAGNOSTIC SND_FEEDER_MULTIFORMAT SND_FEEDER_FULL_MULTIFORMAT SND_FEEDER_RATE_HP SND_PCM_64 SND_OLDSTEREO Manual page updates are on the way. Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
static u_int32_t
screamer_setrecsrc(struct snd_mixer *m, u_int32_t src)
{
return (0);
}
static int
davbus_attach(device_t self)
{
struct davbus_softc *sc;
struct resource *dbdma_irq, *cintr;
void *cookie;
char compat[64];
int rid, oirq, err;
sc = malloc(sizeof(*sc), M_DEVBUF, M_WAITOK | M_ZERO);
sc->aoa.sc_dev = self;
sc->node = ofw_bus_get_node(self);
sc->soundnode = OF_child(sc->node);
/* Map the controller register space. */
rid = 0;
sc->reg = bus_alloc_resource_any(self, SYS_RES_MEMORY, &rid, RF_ACTIVE);
if (sc->reg == NULL)
return (ENXIO);
/* Map the DBDMA channel register space. */
rid = 1;
sc->aoa.sc_odma = bus_alloc_resource_any(self, SYS_RES_MEMORY,
&rid, RF_ACTIVE);
if (sc->aoa.sc_odma == NULL)
return (ENXIO);
/* Establish the DBDMA channel edge-triggered interrupt. */
rid = 1;
dbdma_irq = bus_alloc_resource_any(self, SYS_RES_IRQ,
&rid, RF_SHAREABLE | RF_ACTIVE);
if (dbdma_irq == NULL)
return (ENXIO);
oirq = rman_get_start(dbdma_irq);
DPRINTF(("interrupting at irq %d\n", oirq));
err = powerpc_config_intr(oirq, INTR_TRIGGER_EDGE, INTR_POLARITY_LOW);
if (err != 0)
return (err);
snd_setup_intr(self, dbdma_irq, INTR_MPSAFE, aoa_interrupt,
sc, &cookie);
/* Now initialize the controller. */
bzero(compat, sizeof(compat));
OF_getprop(sc->soundnode, "compatible", compat, sizeof(compat));
OF_getprop(sc->soundnode, "device-id", &sc->device_id, sizeof(u_int));
mtx_init(&sc->mutex, "DAVbus", NULL, MTX_DEF);
device_printf(self, "codec: <%s>\n", compat);
/* Setup the control interrupt. */
rid = 0;
cintr = bus_alloc_resource_any(self, SYS_RES_IRQ,
&rid, RF_SHAREABLE | RF_ACTIVE);
if (cintr != NULL)
bus_setup_intr(self, cintr, INTR_TYPE_MISC | INTR_MPSAFE,
NULL, davbus_cint, sc, &cookie);
/* Initialize controller registers. */
bus_write_4(sc->reg, DAVBUS_SOUND_CTRL, DAVBUS_INPUT_SUBFRAME0 |
DAVBUS_OUTPUT_SUBFRAME0 | DAVBUS_RATE_44100 | DAVBUS_INTR_PORTCHG);
/* Attach DBDMA engine and PCM layer */
err = aoa_attach(sc);
if (err)
return (err);
/* Install codec module */
if (strcmp(compat, "screamer") == 0)
mixer_init(self, &screamer_mixer_class, sc);
else if (strcmp(compat, "burgundy") == 0)
mixer_init(self, &burgundy_mixer_class, sc);
return (0);
}
static void
davbus_cint(void *ptr)
{
struct davbus_softc *d = ptr;
u_int reg, status, mask;
mtx_lock(&d->mutex);
reg = bus_read_4(d->reg, DAVBUS_SOUND_CTRL);
if (reg & DAVBUS_PORTCHG) {
status = bus_read_4(d->reg, DAVBUS_CODEC_STATUS);
if (d->read_status && d->set_outputs) {
mask = (*d->read_status)(d, status);
(*d->set_outputs)(d, mask);
}
/* Clear the interrupt. */
bus_write_4(d->reg, DAVBUS_SOUND_CTRL, reg);
}
mtx_unlock(&d->mutex);
}